In a communications network, a popular method of communicating information involves sending data in packets. A datastream can be segmented into packets and sent across the network to be reassembled at a receiving end. Each packet generally includes a header and a payload. The header includes information related to its destination and reassembly. The payload is a portion of data that recreate the substance of the datastream, which may compose a telephone call, a video clip, a sound recording, a data file, an interactive game, or anything else that existed at the sending end.
Certain factors can diminish the performance of a communications network; prolonged round-trip times and out-of-sequence packets are two examples. Enhancing and/or repairing a network is difficult to do without performance measurements, or metrics. One such metric is an accurate estimation of the Round-Trip Time (RTT) of a data (often in the form of packets) traveling between a sender and a receiver. The RTT of a connection is the time required for a data packet to travel from a source to a receiver and for its acknowledgement to travel back to the source. Techniques that have been used to measure the RTT of connections are relatively inaccurate, resource-intensive to implement, and may affect network performance.
RTT measurements are made with respect to the sender. But making measurements directly at the sender generally requires that some component be placed at the sender to make measurements. Thousands of senders necessitate thousands of measuring devices. To have a representative measure of RTT in a complex network, a great deal of resources would be needed to install the measuring components. This can be expensive. An alternative approach would be to use active techniques. Active-measuring techniques require that artificial packets be injected into the network to be measured. The best that can be expected from an active-technique method is to measure the RTT of those (that is, artificially injected) packets, as opposed to actual, in-use packets going to real customers. All active-measurement techniques suffer from the shortcomings listed above.
Two recent efforts attempt to estimate the RTT at a point along the sender-to-receiver path: the triple-handshake attempt and the slow-start attempt. But these techniques merely compute one RTT sample per TCP connection, making them inaccurate. A sample of the RTT is taken either during the triple handshake or during the slow-start phase of the connection. The primary drawback of these techniques is that they measure only one (typically the first) RTT sample per connection. If the connection experiences variable delays (as such is typical and will be explained with reference to FIG. 10 below) during data transmission, the single measurement take may be highly unrepresentative of actual RTT values associated with that connection. Although the first packet may experience a certain delay, the likelihood of that delay being representative of RTT values over the connection's lifetime is low. This would be somewhat analogous to assuming that the time to commute to work is always the same as the time it took to commute on the very first day. Moreover, there is no guarantee that a particular sample was itself an accurate RTT estimate at the instant it was computed. For example, many Web servers respond to the first packet of a connection with a corresponding acknowledgement packet (ACK) only after some delay. Accordingly, these methods tend to lead to an overestimation of actual RTT times.
The current state of the art could be improved by providing a method and system that accurately measures the RTT of in-use data flowing though a connection in a communications network. The art could also be improved by providing a method for identifying respective packets that compose a data stream.